Jason Lee

About this time last year we were preparing to open our campus in Galesburg, IL.  Since the opening last spring this Campus has been using the recording of Saturday night’s teaching during their 11 am service on Sundays.  This solution has been a fairly stable, but has hasn’t operated without issues.  Additionally our pastor has really wanted to be able to teach the Galesburg campus live but we have multiple limitations… the distance between campuses is 50+ miles, our Peoria Campus is about 3-5 miles from any internet connections that can provide more than 10mb upload and any ISPs offering more speed has wanted nearly 6 figures in construction costs, and point to point connectivity is way more than we can afford. 

In addition to the current limitations the locations we are evaluating for future campuses don’t improve limitations on the list above.. in fact, they might be even more challenging.  Yet, being live is a huge desire from our leadership, so our quest continues.

Since Fiber isn’t an option at either of our campuses (but hopefully soon), we are limited to 50×10 cable modem in Peoria and a 16×2 cable modem in Galesburg.

our requirements for site to site streaming needs to:
-  provide 1080i display in the remote locations
-  not rely upon a private point to point connection
-  not require more than 10 mb upload from the sending location
-  be a solution that is easily reproduced for future sites in smaller towns with limited connectivity
-  be easily powered up and down by volunteers (not a 30 step process between multiple platforms).

We have demoed the Haivision Mako encoder/decoders and while the encoded video they produce is pretty amazing.. the pricetag is way to high to “not break the bank” not to mention  too high for for a “test environment” as we figure out what “live streaming” really means to our organization.  (However, you should at minimum demo their gear.. the Haivision gear gives a great benchmark for anything else you test.)

So we have been doing some testing with various other streaming solutions and thought we might share our mileage.

For our testing / phase 1 project we decided to try several pieces of gear:
-  Marshall VS-102 Encoder/Decoders
-  Wirecast and Wowza Streaming to a Roku
-  Marshall VS-102 and Zixi.com Hybrid

Our Media director thru the Church Technical Leaders and our peers at Willow Creek came across a encoder/decoder hardware (Marshall VS-102) made by the Display Monitor company Marshall Electronics (and Marshall USA).  We had heard that people were having good success using the VS-102 on the LAN but the device was capable of WAN streaming site to site.  The hardware is also able to additionally stream bi-directional Audio… (hmm maybe ClearCom in addition to the video’s audio?) This device across a LAN some pretty awesome results!  I came into the test expecting YouTube quality and was amazed.  If you are looking for a way to extend your HD/SDI video infrastructure this is a device you should checkout.  I don’t know of many hardware encoder/decoders in this price point … let alone something that can provide such a quality signal.

After a local LAN test, We quickly configured the boxes and streamed from site to site over our hardware VPN connections.  Remember we are using cable modems for our internet.. and the streaming at 1mb was solid.. but video quality was lacking… moving much above 2.5 mb we started to get a lot of jitter and audio drop outs.  If you have more than 10 mb upload.. I suspect you would have much better results, but those are just suspicions since we weren’t able to do such testing.

Next enter Chris Kehayias and his teaching us about Zixi.com.  Zixi is an internet based “private CDN” (Content Delivery Network), their strength is delivering HD video content over the public internet, including higher latency connections without the receiving end dropping frames or loosing quality or dropping audio.  The really awesome piece of the puzzle is the ability to stream from Zixi to a Netgear 550 Media Player.. (a endpoint and decoder for under $100 similar concept to roku).

So with all this new knowledge we started some field testing in Galesburg, so I thought I would share what we have tested and what our results were.

We first started streaming site to site with the two VS-102 units, with similar results to our pretesting, dropping frames and audio if we went above 3 mb.  Next we tested a roku streaming via the Amazon EC2 services but had stability issues even at 1 mb.  The quality of the video, when stable was pretty good, but couldn’t get it dialed in to keep a constant connection.  Next we configured the VS-102 encoder to stream a TS-Mpeg stream rather than the default streaming VS-102 to VS-102.  It streams to the Zixi “sending” application on a PC on the same network.  This PC is responsible for applying the Zixi goodness to the stream and sending it to their cloud.  Then at the remote campus we configured a Netgear 550 Media Player, pointed it to the stream and we have video.  We let the stream ‘chew’ for over 2 1/2 hrs, never dropped a frame or received the audio garbled. 

The only real test we couldn’t get working was to us the VS-102 decoder rather than the Netgear 550.  This was because we couldn’t get the VS-102 to receive what the Zixi receiver was pushing across the LAN.  We are working with Marshall support and expect to test this part soon.  Our motivation to getting the VS-102 to be the end point in Galesburg, 1 is the output of HD/SDI but also hopefully an improved video output beyond the Netgear Media Player.

After a fairly strong showing in Galesburg on Friday, we took the advice of Chris Kehayias, testing the stream during service to see the impact when our Wi-Fi is most populated and everything is buzzing… So during the Sunday Am services we tried streaming from the Peoria Campus to my house via Zixi, first 2 hrs total fail.. too much chewing thru our upload… and after smacking around a dropbox upload we were able to get a stable connection to Zixi and from there smooth sailing… even while the Sending Zixi software reporting having to recover over 50k dropped packets.  On the receiving end, you wouldn’t have known that Zixi was working so hard to keep the stream stable.

Overall I have been impressed by the flexibility of the VS-102, however their support has been limited.  The service of Zixi has been pretty amazing..  keeping a stream rock solid even with pretty poor ISP conditions.

Church IT, Hardware, Ministry, Tech CITRT, Galesburg, Hardware, Peoria, Roku, Streaming, Zixi

Project Scope
Preparing for Deployment – Research and Education and Pricing
Deployment of Standard Server & Director Role
Deployment of Edge and Reverse Proxy
Deployment of Lync Voice Capabilities
Configuring Lync PSTN Calling thru Avaya IPOffice
Configure Lync 4 Digit Extension Dialing without DIDs
Configure Asterisk as a SIP Proxy for Avaya IPO and and Lync
Deployment of Lync Client to users
Testing Configuration of Backup Registrar
Training

Continuing the series in our Lync Deployment.  As we are approaching the date that we will completely cut over all users to lync we wanted to build in some redundancy to our deployment. 

We have done this by licensing a second standard server and configuring it in the topology as a backup registrar.  This will allow us to have a fail over server to host all voice calls in the event of a failure to the primary standard edition server (PSE).  The Backup Standard Edition Server (BSE) will provide voice capabilities and limited IM capabilities in a production down situation of the PSE. 
Note: for calls to be made in a ‘failed over’ scenario backup calling routes will need to be configured for the BSE mediation role as discussed in a future post

So we have configured our backup in the topology (how to in a future post) and configured the failover routes so it is time to test the scenarios.  For our testing we want to confirm that the PSE can fail and we can still make calls to the PSTN and if the PSTN is not available make a call out the analog backup lines.

You will want to review the default setting in your topology to set it to the lowest value possible when testing otherwise this test could take 15-20 minutes depending upon your value selected to fail over to a backup registrar. 

Our test was to remove the NIC from the PSE, the Lync clients will disconnect, attempt to re-connect and after the specified time connect to the BSE as the fail over registrar and make calls via the PRI and Pots lines.

However after configuring a Backup Registrar Lync Clients wouldn’t login during a failed server.  The clients would drop the connection as expected but however, they wouldn’t login to the backup registrar with limited functionality as expected. 

Side note… Kudos to @DHannifin helping figure this one out…
check out our awesome buddy Dustin’s blog: http://www.technotesblog.com/ for lots of Uber good Lync goodness.

Even after changing the fail over time to just 30 seconds, the phone handset endpoints would login and calls could be made, but the Lync client would fail to login.   After some digging in the trace logs we found client that wouldn’t connect that we were getting an unauthorized error because the newly added BSE server wasn’t in the user certificate issued by the server to the client so the Lync client didn’t trust the backup registrar.

The Lync Client uses a certificate for communications with the front end server.  This certificate is not updated very often, in fact the default value to when it will update is 8760 HOURS that’s 365 DAYS!  (A little longer than we wanted to wait for our testing…)

You can use the PowerShell command: Get-CSWebServiceConfiguration
to review the current values of your setting for MaxValidityPeriodHours’

Since we didn’t have a year to wait, there are a couple solutions.
1. Change the default value by using the PowerShell command
Set-CSWebServiceConfiguration but this changes the cert settings for all clients and would require time for replication.
or
2. Delete the certificate on the machine that you are using for testing. This is a little more killing a fly with a sledge hammer, but for this testing appeared to be the best solution.

So in a testing scenario where you don’t want to change the re-issue certificate settings, on the machine you are using to test, simply launch an mmc window add the add-in for certificates and choose to manage users certificates.  Next browse to the personal certificates where you should find a certificate named the SIP URI of the user you are logged in as and it is issued by ‘Communications Server’. Delete the certificate and then restart your Lync Client (exit the application not just log off). 

Note: After deleting the cert, before you re-launch the Lync Client, you will need your primary front end server online so a new certificate can be issued to the client on the workstation.  Otherwise you still will not have valid certificate to connect and since the PSE is offline your client will try to connect to the BSE for which it still doesn’t have a valid cert.

After you re-connect to Lync to the PSE you can then power off the PSE (or remove the virtual nic from the virtual machine as we did.) You will notice the Lync client log off and after your Backup Registrar time out passes Lync will login to the Backup Registrar.  You will know this has happed when you see the Lync client display the red bar indicating limited functionality.

If you have correctly configured a backup call route to your gateway, all voice calling will route out the gateway as if your Lync topology was operating normally.

Note: In an actual failover after you have configured all backup routes a call in progress should stay active even while the Lync Client is going thru its log off/log on process to connect to the backup registrar.  If you are in an active call during this fail over, your call should stay connected, BUT it will disconnect if you hit cancel on the Lync client during the reconnection process.

Church IT, Tech, Uncategorized IPO, IPOffice, Lync, POTs, PSTN

Project Scope
Preparing for Deployment – Research and Education and Pricing
Deployment of Standard Server & Director Role
Deployment of Edge and Reverse Proxy
Deployment of Lync Voice Capabilities
Configuring Lync PSTN Calling thru Avaya IPOffice
Configure Lync 4 Digit Extension Dialing without DIDs
Configure Asterisk as a SIP Proxy for Avaya IPO and and Lync
Deployment of Lync Client to users
Testing Configuration of Backup Registrar
Training

 

This post is a continuation of a series of posts about Lync Deployment. The documentation portion of this project has gotten the back burner, and I need to say that a blogger I am not.. but picking up the documentation of this process is important.

This can be used as a resource to configure an Avaya IPOffice (IPO) 412 (software version 5.0) as a Gateway for a Lync deployment calling the PSTN, with AsteriskNOW as a SIP proxy to resolve disconnected calls when placed on hold or transferred, your mileage may vary. Calls are routed over a SIP Trunk (Session Initiation Protocol) configured between the IPO and Asterisk and Asterisk and the Lync Front End server.

Once we deployed the calling from the PSTN via a PRI from the IPOffice to a SIP connection to the Lync Mediation server we were able to make and receive calls from Lync endpoints, however we quickly noticed that when calls were put on hold or needing to be transferred to another extension the call was simply dropped.  It doesn’t matter if the call was being transferred to a Lync extension or an Avaya extension the call would drop.  The only option to “hold” a call was to mute the call.  If Hold was used the call would disconnect.

After a few days of tracking this down we were able to identify this was an issue that happened every time.  It wasn’t specific to a user or extension.  In fact the Avaya white paper noted this as a known issue.

The issue is documented on the final page: https://devconnect.avaya.com/public/download/interop/OCSR2-IPO-PSTN.pdf

The document notes that calls cannot be placed on mute, nor does the PSTN caller ID pass thru to Lync, these notes however that was not our experience.  Mute and Caller ID worked fine on inbound calls.

We tried several different solutions to resolve this issue.  Our first attempt was routing all calls thru an inGate SIParator.  This is basically a SIP proxy device.  We happen to have one laying around from some testing with a SIP dial tone provider.  This device had worked well with the IPO connecting to SIP Trunks that required authentication with a different authentication handshake than the standard Avaya methods.  However the SIParator did allow to proxy the Avaya to Lync SIP trunk, but didn’t resolve the disconnects when holding or transferring calls.

Next we tried to use a SnomOne software PBX, this had some promise, after configuring the call to forward all calls to the Avaya or Lync (which was a hassle) we found that this resulted in calls connecting but the caller not hearing any of the conversation, or the call would just stop passing audio although it remained connected.  We also found that the SnomOne would keep terminated calls still active and you would have to reset the sessions manually.

Finally we landed on an asterisk installation installed on a virtual machine.  We installed Asterisk now (without the web interface) for simplicity.  Once you configure the two sip trunks (one for Avaya and one for Lync) and build the dial plan to forward all calls from Lync to Avaya and all calls from Avaya to Lync the configuration was basically complete.

Much Credit must go to my great Church IT RoundTable peer Dave Mast (@DaveMast) for his Asterisk Programming help! Kuddos to Dave!

Below are the steps to configure the Avaya and Lync to communicate via an Asterisk Proxy.

Install Asterisk on a machine, (in our case a new VM) and note the IP Address you give the server.  Next configure a new Avaya SIP Trunk and ARS Table. The same steps as noted here, except you need to enter the information of your Asterisk server in step 2 as the ITSP IP Field.

After completing steps 1,2,3 and 4. Complete Step 5 to prepare an incoming call route from Asterisk to the IPO.

Step 6 is basically the same and we repurposed the old ARS table that we created but changed the short codes and features a little. 

Note in step 9 if you have extensions on both IPO and Lync you can’t use variables in your short codes.  This remains true.

After step 10 things change a little so I will document that here.  The information may look very similar to the previous instructions with SIP for IPO and Lync with out a proxy but they are a little different.

Because of how you have to pass calls from Avaya to Asterisk you will need to configure you rARS table a little differently.  Step 10 walks you thru a extension with a DID, that in fact is no different.  But Step 11 has changed. I have quoted the information that hasn’t changed and added what needs to be adjusted for the dialing plan to work with Asterisk.

11. Configure routing for For Lync Extensions without DIDs (as documented here).

An ARS entry will have to be created for each Extension since the IPO cannot use variables in the E.164 formatting of the outbound call and Lync requires the call to come in in the +11235556500;ext=4175 format.

The Asterisk can’t pass the formatting with “;” so we will pass just the 4 digit extension from IPO to Asterisk, and our 4 digit dial plan dialing rule that translates calls TO those extensions from a lync endpoint into +11235556500;ext=4175 format will cause the call to route to the extension when it comes into Lync from Asterisk.

This example extensions 4150-4175 don’t have DIDs but were valid Lync extensions, in order for IPO extensions to call extensions 4150-4175 a short code would be required for 41xx Pointing to the the SIP-Lync ARS Table. (Assuming no other extensions in the 4100 range are homed on the IPO).
Then entries for each extension would need to be added to the ARS table.
Code: 41XX, Feature: Dial (if the IPO has any restricted calls to outside use Dial Emergency)
Telephone Number: +1235556500”ext=4150@192.168.1.100”
Telephone Number: 41N”@192.168.1.100” (the “”s are required to tell IPO that nothing contained in this part of the string is a variable. All extensions in this range can use this variable.

Next you will need to configure Lync to see the Asterisk as a gateway.

1. Configure Lync Call routing to use the Asterisk as a Gateway. This assumes you have enabled users for enterprise voice which is a fairly well documented process: http://technet.microsoft.com/en-us/library/gg413011.aspx
After users are enabled, go to the Topology builder and browse the Standard Server. Check the box for Enterprise Voice

Edit the properties and go to the Mediation Server. Enable Collocated Mediation Server. Define your Listening Ports and click new gateway enter the IP address of the Asterisk and the Port that it is listening for SIP traffic on.

Next associate the Gateway with the mediation server

Publish the Topology.

2. Configure Dial Plan and Trunk. Open Lync Control Panel and go to Voice Routing then Trunk configuration open the newly added Gateway and change the Encryption support level to Optional, Uncheck Media Bypass, Uncheck Centralized Media Processing and Uncheck Enable Refer Support.

3. Add a translation rule to call 4 digit extensions on the IPO via the Asterisk. This allows a normalized call from the Lync server to pass just 4 digits to the IPO so it correctly routes to the extension on the IPO.
Starting Digits: +12355565
Length: Exactly 12
Digits to remove: 8
This rule tells the Lync server to simply pass 65xx to the IPO.

You will also need to create a translation rule to pass all digits without the +
Starting Digits: +
Length: Exactly 12
Digits to remove: 0
This rule tells the Lync server to pass 11 digits to the Asterisk.

4. Create a Call Route. Select New Route and name it and add a description. Leave the Pattern to match the default “*” which matches all calls.

5. Scrolling down select Add for Associated Gateways and select the PSTN Gateway. Do not yet associate a PSTN Usage. But confirm the Gateway is added.

6. Create a Site Voice Policy Choose new and select the site you want to add a voice policy for. Add a Description and enable all appropriate features. Then New.

Associate the route just created in step 6 by hitting select

choose the route.

Go back to Routes and edit the Asterisk PSTN route and scroll to the bottom and Associate the PSTN Usage created.

Commit all Changes.

Configure the Asterisk Box

Finally you need to configure the Asterisk.

1.   First Configure the SIP Trunks
   Login as root to the asterisk server and enter: nano –w /etc/asterisk/sip.conf
   Your configuration should be as follows:
   [General]
   bindport=5060
   bindaddr=0.0.0.0
   tcpbindaddr=0.0.0.0
   tcpenable=yes

   [Lync_Trunk_Name]
   type=peer
   port=5068
   host=0.0.0.0 (where 0.0.0.0 is the ip address of your lync front end server)
   dtmfmode=rfc2833
   context=name-of-lync-context (use what ever name you want)
   qualify=yes
   transport=tcp

   [Avaya_Trunk_Name]
   type=peer
   host=0.0.0.0 (where 0.0.0.0 is the ip address of your ayava IPO)
   dtmfmode=rfc2833
   context=name-of-avaya-context (use what ever name you want)
   port=5060
   Transport=tcp
   Hit Ctrl-X and choose to save

   
   

2.   Next Define your Dial plan to forward all calls.
   enter nano –w /etc/asterisk/extensions.conf
   Your configuration should be as follows:
   [Name-of-lync-context]
   exten => _+1xxxxxxxxxx,1,Dial,(SIP/Avaya_Trunk_Name/${EXTEN},45)
   exten => _+12xx,1,Dial,(SIP/Avaya_Trunk_Name/${EXTEN},45)
   exten => _1xxxxxxxxxx,n,Hangup()

   NOTE:
   Line 1 passes PSTN calls from lync to the PSTN
   Line 2 passes 4 diget extensions dialed from the Lync to IPO

   
   [Lync_Trunk_Name]
   exten => _+1xxxxxxxxxx,1,Dial,(SIP/Lync_Trunk_Name/${EXTEN},30)
   exten => _+41xx,1,Dial,(SIP/Avaya_Trunk_Name/${EXTEN},30)
   exten => _1xxxxxxxxxx,n,Hangup()

   NOTE:
   Line 1 passes PSTN calls and all Lync Extensions WITH DID to Lync
   Line 2 passes 4 digit extensions dialed from the IPO that don’t have a DID.
   
   Exit and Save the configuration

   

   One item to note, the value of 45 is the seconds the phone rings before disconnecting the call.  We had to change the default of 30 to 45 because when someone would call a cell phone FROM Lync via the IPO PRI the call sometimes wasn’t getting to the cell phone voicemail before the 30 seconds and would drop the call before the Lync caller could leave a voicemail for the person they were calling.  After adjusting this value above 30 these dropped calls stopped happening.

3. Reload the Configurations
   Enter: asterisk –r
   Enter: reload

   After the config reloads enter: /sip Show peers
   your status for both SIP trunks should show “OK”

   You are new ready to make calls from lync to the PSTN and place calls on hold.

Church IT, Tech Asterisk, Avaya, Call Hold, Call Transfer, IP Office, IPO, IPOffice, Lync, VOIP

A upcoming trip has had me exploring cost effective ways to make traditional phone calls from my mobile device over a Wi-Fi connection.  My trip’s location will be where there is little or no CDMA cell phone coverage and if there is any coverage, Sprint’s rates are fairly expensive.  And since my primary phone is a HTC Evo we need an alternative.

Since most hotels have Wi-Fi or you can usually find a fairly cost effective internet café. The quest for the ability to call any US landline or mobile phone from my mobile device when there is Wi-Fi available has begun.

An alternative is needed since Google voice simply re-routes your calls using GV still uses minutes on a mobile phone as well as requires phone service from your carrier. (Calling from GV redirects the call their phone number and then routes the call from GV to the person you are calling…)

I have found no direct SIP provider that offers free calling to the PSTN (Public Switched Telephone Network), but was able to find a SIP provider that allows free incoming calls… Enter GV Call Back, SipDroid, and SipGate and Google Voice… with those combined you have Free SIP calling anywhere you have a Wi-Fi connection.  Not to mention inbound calling from anyone who has your Google voice number.

Here is the basics:
Using an Android application called Google Voice Call Back you can initiate over an internet connection a Google voice call.  Google Voice then calls you back on your SIP line which then alerts your phone.  Once you answer the SIP call on your mobile device, Google Calls the person you want to talk to, and you are connected via your device on Wi-Fi to someone on their telephone (mobile or Landline).

Here’s how you set it up:

1. Download and install Google Voice Call Back
2. Download and install SIPDroid 
3. Setup a Free Sipgate One Account with SipGate (60 Free outbound minutes and unlimited incoming calls, but you won’t be using any of the outbound calling minutes so it really doesn’t matter)
4. Acquire a “local” US number from SIPGate by entering your zip code.  It doesn’t matter if this number isn’t a local number for you since you won’t be calling this number nor with anyone else.
5. Login to your Google voice account and go to Voice Settings. 
6. Add an additional number and enter your newly acquired SipGate telephone number. (you will be prompted to verify your new Google voice number, but a few more steps need completed first)
7. Back at your SipGate Dashboard, go to settings and then Click on “Voicemail, Call Forwarding &Hunting” and delete the forwarding settings.
   (this will allow for the Google voice call to ring your phone without SipGate voicemail picking up the call before you do on your mobile device)
   
8. Go to “Phone” in the settings of your SipGate Account, Mouse over your IP Phone and select “Sip Credentials” 
   
9. Note the registry, SIP-ID and SIP-Password as you need those in the next steps.
   
10. Launch SipDroid on your phone and press menu and Go to settings
    
11.   Select the first “SIP Account” (Line 1)
     
12. Enter your SIP-ID as the Authorization Username and enter your SIP-Password as the Password.
13.   Select server  or proxy and change from pbxes.org to sipgate.com (leave all other settings as the defaults)
    
14. Scroll down and select which networks SipDroid can use.

    

15. Launch the GV Call Back application and Set “When to use call back” to either use for all calls or ask for every call.
16. Enter your Google Voice username and password.
17. Set the Callback number to your sipgate number.
18. Select phone type as mobile.  Apply Settings.

    

19. You have now configured GV Call Back, SipDroid, and SipGate and Google Voice. 
20. Launch SipDroid and wait for the Yellow indicator to turn Green in the Status Bar.  After the indicator turns green you are able to answer SIPDroid Calls. 
21. Go Back to the Google Voice Settings page and initiate the test call to validate your SIPGate Number.  Your Android Device should begin ringing. Hit the keypad button and enter the code on the dial pad.
    
22. Once your number has been validated, you are ready to make Calls.  With the Google Voice Call back application enabled, and SIPDroid running, go to the phone dial pad and make a call.  GV Call back will indicate it is making a connection

    

23. A few seconds later you will notice the Green handset in the status bar and then the following screen will display.   The first number is your SIPGate Number from which you are receiving the inbound call, the second number is the caller ID of your Google Voice number. (in the case that someone is calling your Google voice number, this line will display the caller ID of the person calling your Google Voice Number.)

    

24.   During the call you will see a screen similar to the incoming call (with the addition of the dial pad icon to enter any touch tones during the call)

    

25.   Once the call is ended the following screen will display and you can resume normal usage of the device.

    

Tech SIP WiFI

After a great morning session (see Session 1 and Session 2 for Notes for the details of the info presented in the Road Show Presentation) at the the Sonicwall Road show presentations we headed over to the Indy Motor Speedway for the lunch and afternoon festivities.

Sonicwall provided lunch in one of suites above the track near the start/finish line.  There was an awesome spread of food and great conversation.

After lunch we headed to the Garages and Pit Road during the Firestone Indy Lights Qualifying.  We had a great time checking out the cars of the Indy Cup drivers as well as the Indy Lights.

Here are some photos from the afternoon… go here for all the photos.

 
 
 
 
 

Church IT, ChurchIT RoundTable, Tech SonicWall

So this week I updated to the most recent versions of Windows Live Writer and Windows Messenger…  One cool feature that I was completely unaware of until I left the house and forgot to disconnect my laptop from Windows Live messenger and expected to get the error “you are logged on to another computer” when i connected on my desktop but didn’t.   I saw something new on the top of the application window reading “available in 2 places”.

You can now be logged into Messenger in two places at once.  This can be a mobile device or a laptop and desktop. The features are noted here

This removes the need to have two Live Messenger logins for multiple places…You can be online both places at once and even continue conversations from the other device or location…. Start in the office, continue the conversation on the Windows Mobile device and finish the conversation on the laptop at home. From the Live Messenger Site: “If you receive a message while you’re signed in on multiple devices, the message will appear on each device that you’re signed in on. Also, if you perform an action on one device, such as open or close a conversation window, the action will occur on all devices.”

 

Really Cool!

Tech

Our Ministry Partnership with ACS has had their scheduling application locked its sights for almost a year now.  Working on a weekly basis their team developing and going live with the product last fall.  Well finally we are live on campus with Facility Scheduler. While we did have some heart burn rolling out the application the overall consensus is that Facility Scheduler is proving to a great reliable tool.   The start of the new calendar year was our date that we selected to migrate away from multiple calendars.

For years our ministry has struggled with global ministry calendaring and FS has been a great help to remove heart burn for our staff when trying to schedule ministry events.  Briefly here is a list of what we combined into on location when Facility Scheduler went online:

• Master Calendar
  • We had an outlook calendar that was basically a glorified 10000 ft overview of what was happening in our ministry without many details and often not updated after things changed in our global planning meetings.
• The Ministry Scheduler
  • This was the predecessor to Facility Scheduler and was used by our Campus Services team to schedule equipment, rooms and other resources. (yes TMS Could have done more i know.)
• Personnel Rotation Calendars
  • We had multiple calendars in crazy places (those specific calendars will remain nameless to protect the innocent individuals who inherited those calendars in crazy places); one in Publisher, one in excel and one in a hybrid of Excel and Outlook.

Now all that data has been entered/migrated into Facility Scheduler (iIoften remind the team at ACS that their new product does far more than schedule the facility and it needs a tune up on a better name… I know they needed to differentiate the new product from the OLD TMS but it was really a better name….maybe we’ll have a naming contest later.)  Anyway… Now our staff can go to one application and view when their events, or personnel are scheduled.

One feature I like about FS is the granular security, we can grant the appropriate permissions to user groups for scheduling specific resources.  In our case each administrative assistant can schedule their own conference room and other area specific resources or personnel without submitting an event request form.  So ministries who "own" a resource can schedule that resource without the hassle of "requesting" to use it.  This is really helpful since now these ministry "owned" resources can be viewed globally when planning large scale events as well as individual ministry needs.

While I am thrilled with the progress we have been able to make I am patiently anticipating some of the development that is on the horizon for Facility Scheduler including:

• Scheduling Requests via  workflow that will get the approval of multiple departments for an event to take place
• Viewing the Calendar (read only) from a view in Outlook
• Management of event registrations that are being processed in  Access ACS (or Our branding Northwoods.me) 
• When you are setting up all the information for the event you can click on a button and configure the online event registrations for that event’s participants.
• Making Requests via an Outlook Meeting Request plug-in
• Displaying the events in Facility Scheduler on our close circuit TV monitor facilitated by a Facility Scheduler ‘Add-On" (development project name "BroadCast")
  • If you haven’t heard anything about this product leave me a note and we’ll get  you connected with the appropriate people this is proving to to be a really really sweet tool!   For those of you using FS and also for those of you who aren’t ACS customers!!!!
  • I am super exited about this one, it has been one of my soap box items since day one….  The data is already in our scheduling applications why isn’t there a tool to display it.  This tool looks like it will support multiple data sources not just ACS facility scheduler.
• Auto-magically generating pages on the fly for each event so that events and registration links can be published via the web without manually setting up those pages (all available to the public without the need for knowing any credentials)
• Alerts telling the FS admin that Online Registrations are nearing capacity or are full to aid in selecting a location for a specific event.
• Alerts Reminding the event planner that their event is scheduled and what resources are included a few days before the event.

 

I know others of you are also diving into Facility Scheduler and I would be interested in hearing your success so far as well as your heart burn.  Anyone interested in an online roundtable discussion about FS?

I know several are in the process of moving to Facility Scheduler you might check out their blogs for more information too:

Shawn Ross’s blog

Jeff Suever ’s blog
Jeff has been posting a bit about his experience with Facility Scheduler too.  I would also weigh in to our friends at ACS that Jeff is absolutely on the right track FS needs to be able to accommodate Links to an event page outside the ACS generated pages and would push a little more than Jeff to say the event description editor needs to be able to natively accommodate hyper links rather than dropping code into the description (one because this might do wonky things to "BroadCast").  Also Jeff has noted that the individual event page links expire after 24 hours… and while I understand the security that is in play, we need to be able to have a link for each event that doesn’t expire after 24 hours.  Additionally I’ll add it would be nice to be able to configure what displays on these pages…. some may want some data to appear others won’t (example you might want to toggle on or off the "confirmed" line to the general public.

Overall Facility Scheduler is a great product and it is moving forward a great speed.  If you are an ACS customer and haven’t looked into Facility Scheduler don’t miss out…. and if you are looking for a Scheduling application And AREN"T an ACS customer this should be one of the applications on the top of your list.

Church IT, Tech

In a recent series of posts I documented the process that we used to deploy Vista in our test scenarios and then to several Dell 755 machines that are now in production… Problem with those posts, the links from the first post to the subsequent posts were wrong.  Those links have now been updated and added below incase you had issue navigating thru the documentation.

 

• Vista Deployment
• Installing and Configuring MDT
• Deploying Vista to VMware Test box and Dell 755
• MDT and Dell 755 Network issues
• Deploying to Vista 64 bit to Dell 755

Church IT, Tech

I have noted when tech support has performed below our expectations in the past so that means I should also note when we receive excellent support too, right?

During our MDT Deployment I had had enough and decided that I wasn’t going to spend any more time on trouble shooting the issues… Thankfully we recently purchased a MSDN subscription and that included 4 telephone support incidents so Jeremie and I decided to give it a try.  I called the 800 number and was quickly routed to Tanner S. a senior support engineer who specializes in MDT and WDS.  I stated the issues we were having and Tanner quickly resolved our issue and I was on our way to success… so I thought.

After I completed the call with Tanner we came upon several other issues so I responded to the email he had sent me with the case number and asked 2 more questions which he quickly resolved.

After those issues were resolved we found a couple more problems and we contacted Tanners backup Chuck W.  who helped us resolve those issues.

These guys helped us resolve multiple issues which included:
- Windows PE not loading
- How to edit the startnet.cmd file when we had a timeout issue connecting to the WDS host
- Drivers not getting installed on images
- Including Intel Chipset drivers that are packaged and aren’t included in the OOBdrivers
- Application Installation errors during the MDT install

So needless to say these guys went over an above to get us going.  They could have simply said that the subsequent questions were not part of the case and closed the case but they were willing to conceder our support request all encompassing from the setup of MDT to the final deployment to our Dell 755s and we were allowed to ask questions along the way.  They solved our 1st problem and multiple other smaller issues afterward.  They both did it in a very gracious, and kind demeanor.  While they both could have told us to go find the answers on live.com or to read the documentation they were willing to answer all our questions… Even staying after their shift ended to make sure the test deployment completed without any errors after we made changes to the boot images. 

Kudos to Tanner and Chuck… Great customer support and a job well done!!!

Church IT, Tech

After we were able to get the deployment of Vista 32bit out to our Dell 755s tackling the install of Vista 64bit was next on the agenda.  The primary reason for pushing out 64bit was because of the memory threshold limit on 32bit.  Several of our old Dell 740s had 2 gb of added memory we wanted to move over to our new 755 boxes which put us over the 3.5gb limit.  Not a huge issue, but I also new down the road we would want to deploy 64bit versions of Server 2008 so we might as well work out the issues now.

Before you can start a 64bit install you have to add a 64bit OS to the list in MDT.  This is the same process as adding a 32bit OS and creating the task sequence as documented in the first MDT post.  When you include your 64bit OS in a task sequence, be sure in the title and description to note that this is the 64bit installer so there is no confusion later. After the OS and Task Sequence are added you will need to add any 64bit Out of the Box Drivers this is the same process that you did with your 32 bit drivers.

Lastly you need to make 64bit a supported platform with your deployment point.  Up to this point when you updated the boot image MDT only updated the 32bit .wim file so now you need to tell MDT to also update the 64 bit boot image.

Go to Deployment Point, and Choose Properties for the Deployment Point and then on the general tab tick the check box next to x64 and choose OK.  Lastly update your Deployment Point.

 

After your deployment point is updated you will need to add this new boot image to your WDS server.  Go to Server Manager and navigate to the boot images in your WDS server.  Right Click on Boot images and select new boot image.  This time you will select the 64bit boot image, LiteTouchPE_x64.wim, found in:
\Distribution\Boot\.  You can leave both the 32bit and 64 bit boot images enabled so when you pxe boot off your server you select the appropriate architecture for your install.

When you PXE boot off the WDS Server you will be presented with the two LiteTouch boot images, select the x64 image.

After you select the x64 image the WindowsPE installer that you have seen in the 32 bit installs will display… this time with one exception, the Operating systems displayed as tasks available are the 64bit options you added in the task sequences.

 

64 bit OS Install “Got-Ya’s”

1.  The intel sata driver for the Dell 755 appears in MDT as a driver that is both 32 and 64 bit.  Its not.  When you boot into WindowsPE the first time after adding all your drivers and updating the boot images you will get a lovely error like the following:
File: \windows\system32\ddrivers\iastor.sys
Status: 0xc0000359
Info: Windows Failed to load because a critical system driver is missing or corrupt.

This happens only after you update the 64bit boot image from MDT.  The original 64bit boot image has the needed storage controller driver but when you update MDT it includes the OOB drivers you added.  Since intel’s storage driver is really not a 64 bit driver like MDT thinks it is the 64bit WindowsPE bombs.   So what do you need to do?  The easiest way to do fix this is the following steps:

• Delete all drivers from the Out of Box Drivers in MDT
• Download both the 64 bit and 32 bit Intel Matrix Storage Manager Drivers.
• Add only the 32 bit Driver to OOBD
• Open the properties of that driver and un-tick the x64 check box.

• Next add the 64 bit driver.  This time you will have to select “Import drivers even if they are duplicates of an existing driver”

• Next you will need to edit this newly added driver.  The easiest way to find this driver is to sort all OOB Drivers by Platform.  The newly added driver will display x86, x64.  Edit that driver to only support x64 platform.

• After your drivers are updated, import all your drivers again and update your deployment point.

 

See other Posts on Vista Deployment with MDT:

• Vista Deployment
• Installing and Configuring MDT
• Deploying Vista to VMware Test box and Dell 755
• MDT and Dell 755 Network issues
• Deploying to Vista 64 bit to Dell 755

Church IT, Tech